I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. Most calls are fine, all incoming calls are fine, but I am getting echo on a significant number of outgoing calls. The person on the other side hears a perfect call, but the SIPphone side gets to hear themselves. It happens 100% of the time to some numbers (outgoing only), and only sporadically to others. Has anyone ever experienced this? the RTT to the phones from the server is less than 10ms and it is a 100mbit network with no traffic and cisco switches. zapata.conf attached below: Note: The commented out gain of +2 on outgoing seems to make no difference to the effect. Has anyone got any ideas? ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] group => 1,16 [channels] spanmap => 1,1,1 language=en context=from-pstn rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 ;txgain=2.0 txgain=0.0 rxgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no pridialplan=unknown overlapdial=yes signalling=pri_cpe switchtype=euroisdn channel=> 1-12 faxdetect=both
yes it sounds very familier. I would suggest not tweaking settings for gain on a T1, it should work fine at 0db. I got tired of chasing down these issues with our/other carriers because the software echo cancellor on the system is just not capable of performing this task. I suspect the new cards with the onboard DSP will be much improved. I broke down and bought dedicated gear to handle and it has been flawless - zero complaints since installed. But if you go this route get your checkbook out, they are not cheap unless you go the ebay route. As a side note the echo gear has been keeping a running tab on the worst connections on each of the channels. per the asterisk docs "Accordingly the number of taps equate to a 2ms, 4ms, 8ms, 16ms or 32ms tail length." or a maximum of 32ms cancelling. In a good world this would be sufficient. From my measured stats you can see where this is not even close to doing the job. This particular hardware performs upto 192ms with no problem. Good Luck CHANNEL 1 2 3 4 5 6 WORST DLY -- -- -- -- -- -- TIME -- -- -- -- -- -- DATE -- -- -- -- -- -- CHANNEL 7 8 9 10 11 12 WORST DLY -- 9 ms 10 ms 10 ms 25 ms 21 ms TIME -- 14:21:51 11:20:26 15:45:22 10:43:54 13:22:52 DATE -- 07/13/05 06/15/05 07/19/05 07/18/05 06/07/05 CHANNEL 13 14 15 16 17 18 WORST DLY 27 ms 108 ms 36 ms 159 ms 37 ms 150 ms TIME 14:33:13 10:15:25 15:09:07 11:30:02 10:41:32 15:09:07 DATE 07/13/05 06/28/05 07/06/05 07/14/05 07/11/05 07/07/05 CHANNEL 19 20 21 22 23 24 WORST DLY 123 ms 168 ms 135 ms 144 ms 184 ms -- TIME 13:30:32 13:22:28 08:05:15 12:02:27 07:47:07 -- DATE 05/27/05 06/23/05 06/02/05 06/24/05 05/31/05 -- On Aug 4, 2005, at 4:10 PM, Robbie Hughes wrote:> I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. > Most calls are fine, all incoming calls are fine, but I am getting > echo on a significant number of outgoing calls. > The person on the other side hears a perfect call, but the SIPphone > side gets to hear themselves. > > It happens 100% of the time to some numbers (outgoing only), and > only sporadically to others. > > Has anyone ever experienced this? > the RTT to the phones from the server is less than 10ms and it is a > 100mbit network with no traffic and cisco switches. > > zapata.conf attached below: > Note: The commented out gain of +2 on outgoing seems to make no > difference to the effect. > > > Has anyone got any ideas? > > ; > ; Zapata telephony interface > ; > ; Configuration file > > [trunkgroups] > group => 1,16 > [channels] > spanmap => 1,1,1 > language=en > context=from-pstn > rxwink=300 ; Atlas seems to use long (250ms) winks > > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > echotraining=800 > ;txgain=2.0 > txgain=0.0 > rxgain=0.0 > > group=1 > callgroup=1 > pickupgroup=1 > immediate=no > pridialplan=unknown > overlapdial=yes > signalling=pri_cpe > switchtype=euroisdn > channel=> 1-12 > faxdetect=both > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Robbie: I fought with echocancel and various parameters for a long time with little luck. Then I uncommented AGGRESSIVE_SUPPRESSOR and DISABLED the Fax/tone detection in in zconfig.h since we're not faxing via Asterisk. Recompiled and all echo disappeared. Hope that helps. -Rob -- Robert Goodyear Brand Up LLC http://www.brand-up.com On Aug 4, 2005, at 2:10 PM, Robbie Hughes wrote:> I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. > Most calls are fine, all incoming calls are fine, but I am getting > echo on a significant number of outgoing calls. > The person on the other side hears a perfect call, but the SIPphone > side gets to hear themselves. > > It happens 100% of the time to some numbers (outgoing only), and > only sporadically to others. > > Has anyone ever experienced this? > the RTT to the phones from the server is less than 10ms and it is a > 100mbit network with no traffic and cisco switches. > > zapata.conf attached below: > Note: The commented out gain of +2 on outgoing seems to make no > difference to the effect. > > > Has anyone got any ideas? > > ; > ; Zapata telephony interface > ; > ; Configuration file > > [trunkgroups] > group => 1,16 > [channels] > spanmap => 1,1,1 > language=en > context=from-pstn > rxwink=300 ; Atlas seems to use long (250ms) winks > > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > echotraining=800 > ;txgain=2.0 > txgain=0.0 > rxgain=0.0 > > group=1 > callgroup=1 > pickupgroup=1 > immediate=no > pridialplan=unknown > overlapdial=yes > signalling=pri_cpe > switchtype=euroisdn > channel=> 1-12 > faxdetect=both > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
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